February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

last git problem ?
by Madovsky 03 Feb '11

03 Feb '11

after every hangup now FS quits unexpeclty,without any error message....

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Voice quality monitoring via loopback on sip endpoint
by Marc De Corny 02 Feb '11

02 Feb '11

Hi all, I am thinking of a way of testing the quality of a voice call. Ideally i would like to create a call from FS with a loopback parameter that tells phone to answer the call automatically and loopback the RTP. This is a common functionality in most phoned (for example cisco) The sent and received packets could be compared for jitter, latency and packet loss and a result extracted. Does this already exist? I know there some specific tool that accomplish this bit they are expensive and awkward to use. Starting and retrieving the result from an api would be fantastic. Any ideas are welcome. Thanks Marc

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Re: [Freeswitch-users] last git problem ?
by Madovsky 02 Feb '11

02 Feb '11

maybe thies unusual log lines after hangup before FS quits 2011-02-02 14:58:35.549213 [DEBUG] switch_ivr_bridge.c:500 sofia/internal/9999999999999@default ending bridge by request from read function 2011-02-02 14:58:35.549213 [DEBUG] switch_ivr_bridge.c:494 sofia/internal/9999999999999@default ending bridge by request from write function nothing else ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, February 02, 2011 2:54 PM Subject: last git problem ? after every hangup now FS quits unexpeclty,without any error message....

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execute_on_answer
by Madovsky 02 Feb '11

02 Feb '11

possible to use execute_on_answer=<api> value; <api> value ??? thanks

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execute_on_ring executing on answer
by François Delawarde 02 Feb '11

02 Feb '11

Hi, Doing some testing with this morning's git (Fri Jan 28) I just found out that the execute_on_ring application runs when the destination answers instead of when it rings. So far, I can't seem to find out the reason. Could it be some configuration issue? Here a call log showing the phenomenon with a simple bridge: <action application="bridge" data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103@192.168.10.22:5060"/> http://pastebin.freeswitch.org/15168 Thanks, François.

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Re: [Freeswitch-users] Route incoming analog calls
by Eric Michel 02 Feb '11

02 Feb '11

Steve thank you for the pointers, you got me going in the right direction. Here is what I came up with (so far it seems to work). <extension name="Line 2(6820)" continue="true"> <condition field="source" expression="mod_freetdm"/> <condition field="${channel_name}" expression="^(FreeTDM/2:2/)$"> <action application="info"/> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="transfer_ringback=$${us-ring}"/> <action application="answer"/> <action application="transfer" data="2000"/> </condition> </extension> All I have to do is modify the second number in the expression. 2:3 line three, 2:4 line four etc. -Eric

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Need to change "identity display" in dialog-info event
by Helmut Kuper 02 Feb '11

02 Feb '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello is there a way to get the value of "effective_caller_id_name" in dialog-info's "remote identity display"? And further: Is it possible to get that value also accessible for display updates, so that after picking up a call the display is updated with the picked up call's effective_caller_id_name? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk1JjOUACgkQ4tZeNddg3dyx7wCffCCNvFCJczlmn9qKwvwXm8Ms cdYAnR7tkUactRxkkqg+9D2h8Ig9TQB2 =ZcKJ -----END PGP SIGNATURE-----

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open g729
by Gustavo Espeche 02 Feb '11

02 Feb '11

Hello, some one can compile open g729 to work with freeswitch? http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ i appreciate a lot if some one has some experience in it. Best Regards. Gustavo Espeche www.easyipcall.com

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How to prevent source channel hangup after uuid_bridge completion (outbound ESL)?
by Dmitry Sytchev 02 Feb '11

02 Feb '11

Hi I'm trying to originate call from outbound ESL and then bridge it to original channel where ESL is launched on. I use api originate with &park(), then uuid_bridge. Everything works as expected, but when second leg hangs up, first leg gets hangup too. How can I prevent first leg from hanging up and continue to process it in ESL? Thanks in advance -- Best regards, Dmitry Sytchev, IT Engineer

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Route incoming analog calls.
by Eric Michel 02 Feb '11

02 Feb '11

I've got a Sangoma A200 with six FXO ports and two FXS. I can call out just fine, but I'm having trouble figuring out how to route incoming calls. We've got five POTS lines and all I've been able to do is route all incoming calls to a specific extension. How do I go about routing individual POTS lines to different groups/extensions? <extension name="analog test2" continue="true"> <condition field="source" expression="mod_freetdm"> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="transfer_ringback=$${us-ring}"/> <action application="answer"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> That is my current conf/public/inbound dial plan. Shouldn't I be able to add a second condition that tests what FXO port is receiving the call and allow me to route it accordingly? If yes, could I get an example? The answer may be really simple, I've just been unable to find it. Thanks, Eric

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